diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index 3457350b9..fac379948 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -309,7 +309,7 @@ srt_server {
 #############################################################################################
 # WebRTC server section
 #############################################################################################
-rtc {
+rtc_server {
     # Whether enable WebRTC server.
     # default: off
     enabled         on;
diff --git a/trunk/conf/rtc.conf b/trunk/conf/rtc.conf
index 300b81d6f..f4578cec5 100644
--- a/trunk/conf/rtc.conf
+++ b/trunk/conf/rtc.conf
@@ -18,7 +18,7 @@ http_api {
 stats {
     network         0;
 }
-rtc {
+rtc_server {
     enabled         on;
     # Listen at udp://8000
     listen          8000;
@@ -32,5 +32,9 @@ rtc {
 }
 
 vhost __defaultVhost__ {
+    rtc {
+        enabled     on;
+        bframe      discard;
+    }
 }
 
diff --git a/trunk/configure b/trunk/configure
index ab9402506..df55f860e 100755
--- a/trunk/configure
+++ b/trunk/configure
@@ -256,7 +256,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
             "srs_app_ingest" "srs_app_ffmpeg" "srs_app_utility" "srs_app_edge"
             "srs_app_heartbeat" "srs_app_empty" "srs_app_http_client" "srs_app_http_static"
             "srs_app_recv_thread" "srs_app_security" "srs_app_statistic" "srs_app_hds"
-            "srs_app_mpegts_udp" "srs_app_rtp" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
+            "srs_app_mpegts_udp" "srs_app_rtc" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
             "srs_app_caster_flv" "srs_app_process" "srs_app_ng_exec"
             "srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
             "srs_app_coworkers" "srs_app_hybrid")
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index b79fdadd7..7ec283f5f 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -3486,7 +3486,7 @@ srs_error_t SrsConfig::check_normal_config()
             && n != "srs_log_tank" && n != "srs_log_level" && n != "srs_log_file"
             && n != "max_connections" && n != "daemon" && n != "heartbeat"
             && n != "http_api" && n != "stats" && n != "vhost" && n != "pithy_print_ms"
-            && n != "http_server" && n != "stream_caster" && n != "rtc" && n != "srt_server"
+            && n != "http_server" && n != "stream_caster" && n != "rtc_server" && n != "srt_server"
             && n != "utc_time" && n != "work_dir" && n != "asprocess"
             && n != "ff_log_level" && n != "grace_final_wait" && n != "force_grace_quit"
             && n != "grace_start_wait" && n != "empty_ip_ok" && n != "disable_daemon_for_docker"
@@ -3673,7 +3673,7 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "play" && n != "publish" && n != "cluster"
                 && n != "security" && n != "http_remux" && n != "dash"
                 && n != "http_static" && n != "hds" && n != "exec"
-                && n != "in_ack_size" && n != "out_ack_size") {
+                && n != "in_ack_size" && n != "out_ack_size" && n != "rtc") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.%s", n.c_str());
             }
             // for each sub directives of vhost.
@@ -3819,6 +3819,13 @@ srs_error_t SrsConfig::check_normal_config()
                         return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.bandcheck.%s of %s", m.c_str(), vhost->arg0().c_str());
                     }
                 }
+            } else if (n == "rtc") {
+                for (int j = 0; j < (int)conf->directives.size(); j++) {
+                    string m = conf->at(j)->name;
+                    if (m != "enabled" && m != "bframe") {
+                        return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.bandcheck.%s of %s", m.c_str(), vhost->arg0().c_str());
+                    }
+                }
             }
         }
     }
@@ -4266,13 +4273,13 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
     return ::atoi(conf->arg0().c_str());
 }
 
-int SrsConfig::get_rtc_enabled()
+int SrsConfig::get_rtc_server_enabled()
 {
-    SrsConfDirective* conf = root->get("rtc");
-    return get_rtc_enabled(conf);
+    SrsConfDirective* conf = root->get("rtc_server");
+    return get_rtc_server_enabled(conf);
 }
 
-bool SrsConfig::get_rtc_enabled(SrsConfDirective* conf)
+bool SrsConfig::get_rtc_server_enabled(SrsConfDirective* conf)
 {
     static bool DEFAULT = false;
     
@@ -4288,11 +4295,11 @@ bool SrsConfig::get_rtc_enabled(SrsConfDirective* conf)
     return SRS_CONF_PERFER_FALSE(conf->arg0());
 }
 
-int SrsConfig::get_rtc_listen()
+int SrsConfig::get_rtc_server_listen()
 {
     static int DEFAULT = 8000;
     
-    SrsConfDirective* conf = root->get("rtc");
+    SrsConfDirective* conf = root->get("rtc_server");
     if (!conf) {
         return DEFAULT;
     }
@@ -4305,11 +4312,11 @@ int SrsConfig::get_rtc_listen()
     return ::atoi(conf->arg0().c_str());
 }
 
-std::string SrsConfig::get_rtc_candidates()
+std::string SrsConfig::get_rtc_server_candidates()
 {
     static string DEFAULT = "*";
     
-    SrsConfDirective* conf = root->get("rtc");
+    SrsConfDirective* conf = root->get("rtc_server");
     if (!conf) {
         return DEFAULT;
     }
@@ -4332,6 +4339,48 @@ std::string SrsConfig::get_rtc_candidates()
     return (conf->arg0().c_str());
 }
 
+SrsConfDirective* SrsConfig::get_rtc(string vhost)
+{
+    SrsConfDirective* conf = get_vhost(vhost);
+    return conf? conf->get("rtc") : NULL;
+}
+
+bool SrsConfig::get_rtc_enabled(string vhost)
+{
+    static bool DEFAULT = false;
+    
+    SrsConfDirective* conf = get_rtc(vhost);
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("enabled");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
+bool SrsConfig::get_rtc_bframe_discard(string vhost)
+{
+    static bool DEFAULT = false;
+    
+    SrsConfDirective* conf = get_rtc(vhost);
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("enabled");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return conf->arg0() == "discard";
+}
+
 SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
 {
     srs_assert(root);
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index 768e8368f..e50aff296 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -501,10 +501,14 @@ public:
 
 // rtc section
 public:
-    virtual int get_rtc_enabled();
-    virtual bool get_rtc_enabled(SrsConfDirective* conf);
-    virtual int get_rtc_listen();
-    virtual std::string get_rtc_candidates();
+    virtual int get_rtc_server_enabled();
+    virtual bool get_rtc_server_enabled(SrsConfDirective* conf);
+    virtual int get_rtc_server_listen();
+    virtual std::string get_rtc_server_candidates();
+
+    SrsConfDirective* get_rtc(std::string vhost);
+    bool get_rtc_enabled(std::string vhost);
+    bool get_rtc_bframe_discard(std::string vhost);
 
 // vhost specified section
 public:
diff --git a/trunk/src/app/srs_app_rtp.cpp b/trunk/src/app/srs_app_rtc.cpp
similarity index 90%
rename from trunk/src/app/srs_app_rtp.cpp
rename to trunk/src/app/srs_app_rtc.cpp
index de8932018..26351e4da 100644
--- a/trunk/src/app/srs_app_rtp.cpp
+++ b/trunk/src/app/srs_app_rtc.cpp
@@ -21,7 +21,7 @@
  * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
  */
 
-#include <srs_app_rtp.hpp>
+#include <srs_app_rtc.hpp>
 
 #include <sys/types.h>
 #include <sys/stat.h>
@@ -49,11 +49,13 @@ using namespace std;
 #include <srs_app_utility.hpp>
 #include <srs_app_http_hooks.hpp>
 #include <srs_protocol_format.hpp>
+#include <srs_rtmp_stack.hpp>
 #include <openssl/rand.h>
 
 SrsRtpMuxer::SrsRtpMuxer()
 {
     sequence = 0;
+    discard_bframe = false;
 }
 
 SrsRtpMuxer::~SrsRtpMuxer()
@@ -99,9 +101,10 @@ srs_error_t SrsRtpMuxer::frame_to_packet(SrsSharedPtrMessage* shared_frame, SrsF
             }
 
             srs_verbose("nal_type=%d, slice type=%d", nal_type, slice_type);
-            // TODO: Use config to determine how to process B frame
             if (slice_type == SrsAvcSliceTypeB || slice_type == SrsAvcSliceTypeB1) {
-                continue;
+                if (discard_bframe) {
+                    continue;
+                }
             }
         }
 
@@ -179,11 +182,8 @@ srs_error_t SrsRtpMuxer::packet_fu_a(SrsSharedPtrMessage* shared_frame, SrsForma
         p += packet_size;
         nb_left -= packet_size;
 
-        srs_verbose("rtp fu-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s", 
-            sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType, 
-            srs_string_dumps_hex(stream->data(), 12).c_str(),
-            srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
-
+        srs_verbose("rtp fu-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u", 
+            sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
 
         SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
         rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
@@ -224,11 +224,8 @@ srs_error_t SrsRtpMuxer::packet_single_nalu(SrsSharedPtrMessage* shared_frame, S
 
     stream->write_bytes(sample->bytes, sample->size);
 
-    srs_verbose("sample=%s", srs_string_dumps_hex(sample->bytes, sample->size).c_str());
-    srs_verbose("rtp single nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s", 
-        sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType, 
-        srs_string_dumps_hex(stream->data(), 12).c_str(),
-        srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
+    srs_verbose("rtp single nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u", 
+        sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
 
     SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
     rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
@@ -275,10 +272,8 @@ srs_error_t SrsRtpMuxer::packet_stap_a(const string &sps, const string& pps, Srs
     stream->write_2bytes(pps.size());
     stream->write_bytes((char*)pps.data(), pps.size());
 
-    srs_verbose("rtp stap-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s", 
-        (sps.size() + pps.size()), sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType, 
-        srs_string_dumps_hex(stream->data(), 12).c_str(),
-        srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
+    srs_verbose("rtp stap-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u", 
+        (sps.size() + pps.size()), sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
 
     SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
     rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
@@ -288,7 +283,7 @@ srs_error_t SrsRtpMuxer::packet_stap_a(const string &sps, const string& pps, Srs
     return err;
 }
 
-SrsRtp::SrsRtp()
+SrsRtc::SrsRtc()
 {
     req = NULL;
     hub = NULL;
@@ -298,12 +293,12 @@ SrsRtp::SrsRtp()
     last_update_time = 0;
 }
 
-SrsRtp::~SrsRtp()
+SrsRtc::~SrsRtc()
 {
     srs_freep(rtp_h264_muxer);
 }
 
-void SrsRtp::dispose()
+void SrsRtc::dispose()
 {
     if (enabled) {
         on_unpublish();
@@ -311,14 +306,14 @@ void SrsRtp::dispose()
 }
 
 // TODO: FIXME: Dead code?
-srs_error_t SrsRtp::cycle()
+srs_error_t SrsRtc::cycle()
 {
     srs_error_t err = srs_success;
     
     return err;
 }
 
-srs_error_t SrsRtp::initialize(SrsOriginHub* h, SrsRequest* r)
+srs_error_t SrsRtc::initialize(SrsOriginHub* h, SrsRequest* r)
 {
     srs_error_t err = srs_success;
     
@@ -326,11 +321,12 @@ srs_error_t SrsRtp::initialize(SrsOriginHub* h, SrsRequest* r)
     req = r;
 
     rtp_h264_muxer = new SrsRtpMuxer();
+    rtp_h264_muxer->discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
     
     return err;
 }
 
-srs_error_t SrsRtp::on_publish()
+srs_error_t SrsRtc::on_publish()
 {
     srs_error_t err = srs_success;
 
@@ -341,6 +337,10 @@ srs_error_t SrsRtp::on_publish()
     if (enabled) {
         return err;
     }
+
+	if (!_srs_config->get_rtc_enabled(req->vhost)) {
+        return err; 
+    }
     
     // if enabled, open the muxer.
     enabled = true;
@@ -351,7 +351,7 @@ srs_error_t SrsRtp::on_publish()
     return err;
 }
 
-void SrsRtp::on_unpublish()
+void SrsRtc::on_unpublish()
 {
     // support multiple unpublish.
     if (!enabled) {
@@ -361,7 +361,7 @@ void SrsRtp::on_unpublish()
     enabled = false;
 }
 
-srs_error_t SrsRtp::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
+srs_error_t SrsRtc::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
 {
     srs_error_t err = srs_success;
     
@@ -391,7 +391,7 @@ srs_error_t SrsRtp::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* forma
     return err;
 }
 
-srs_error_t SrsRtp::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
+srs_error_t SrsRtc::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
 {
     srs_error_t err = srs_success;
 
diff --git a/trunk/src/app/srs_app_rtp.hpp b/trunk/src/app/srs_app_rtc.hpp
similarity index 93%
rename from trunk/src/app/srs_app_rtp.hpp
rename to trunk/src/app/srs_app_rtc.hpp
index 842d82620..2a047d110 100644
--- a/trunk/src/app/srs_app_rtp.hpp
+++ b/trunk/src/app/srs_app_rtc.hpp
@@ -21,8 +21,8 @@
  * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
  */
 
-#ifndef SRS_APP_RTP_HPP
-#define SRS_APP_RTP_HPP
+#ifndef SRS_APP_RTC_HPP
+#define SRS_APP_RTC_HPP
 
 #include <srs_core.hpp>
 
@@ -53,12 +53,15 @@ const uint8_t kEnd   = 0x40;
 // FIXME: ssrc can relate to source
 const uint32_t kVideoSSRC = 3233846889;
 
+// TODO: Define interface class like ISrsRtpMuxer to support SrsRtpOpusMuxer and so on.
 class SrsRtpMuxer
 {
 private:
     uint16_t sequence;
     std::string sps;
     std::string pps;
+public:
+    bool discard_bframe;
 public:
     SrsRtpMuxer();
     virtual ~SrsRtpMuxer();
@@ -70,7 +73,7 @@ private:
     srs_error_t packet_stap_a(const std::string &sps, const std::string& pps, SrsSharedPtrMessage* shared_frame, std::vector<SrsRtpSharedPacket*>& rtp_packet_vec);
 };
 
-class SrsRtp
+class SrsRtc
 {
 private:
     SrsRequest* req;
@@ -80,8 +83,8 @@ private:
     SrsRtpMuxer* rtp_h264_muxer;
     SrsOriginHub* hub;
 public:
-    SrsRtp();
-    virtual ~SrsRtp();
+    SrsRtc();
+    virtual ~SrsRtc();
 public:
     virtual void dispose();
     virtual srs_error_t cycle();
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 6eef83fba..77de89bd6 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -46,7 +46,7 @@ using namespace std;
 #include <srs_app_dtls.hpp>
 #include <srs_app_utility.hpp>
 #include <srs_app_config.hpp>
-#include <srs_app_rtp.hpp>
+#include <srs_app_rtc.hpp>
 #include <srs_app_source.hpp>
 #include <srs_app_server.hpp>
 #include <srs_service_utility.hpp>
@@ -101,7 +101,7 @@ std::vector<std::string> SrsCandidate::get_candidate_ips()
 {
     std::vector<std::string> candidate_ips;
 
-    string candidate = _srs_config->get_rtc_candidates();
+    string candidate = _srs_config->get_rtc_server_candidates();
     if (candidate == "*" || candidate == "0.0.0.0") {
         std::vector<std::string> tmp = srs_get_local_ips();
         for (int i = 0; i < (int)tmp.size(); ++i) {
@@ -189,7 +189,7 @@ srs_error_t SrsSdp::encode(string& sdp_str)
     std::vector<string> candidate_ips = SrsCandidate::get_candidate_ips();
     for (int i = 0; i < (int)candidate_ips.size(); ++i) {
         ostringstream os;
-        os << "a=candidate:10 1 udp 2115783679 " << candidate_ips[i] << " " << _srs_config->get_rtc_listen() <<" typ host generation 0\\r\\n";
+        os << "a=candidate:10 1 udp 2115783679 " << candidate_ips[i] << " " << _srs_config->get_rtc_server_listen() <<" typ host generation 0\\r\\n";
         candidate_lines += os.str();
     }
 
@@ -1179,11 +1179,11 @@ srs_error_t SrsRtcServer::listen_udp()
 {
     srs_error_t err = srs_success;
 
-    if (!_srs_config->get_rtc_enabled()) {
+    if (!_srs_config->get_rtc_server_enabled()) {
         return err;
     }
 
-    int port = _srs_config->get_rtc_listen();
+    int port = _srs_config->get_rtc_server_listen();
     if (port <= 0) {
         return srs_error_new(ERROR_RTC_PORT, "invalid port=%d", port);
     }
diff --git a/trunk/src/app/srs_app_source.cpp b/trunk/src/app/srs_app_source.cpp
index bccf5aae7..2d26c9e33 100755
--- a/trunk/src/app/srs_app_source.cpp
+++ b/trunk/src/app/srs_app_source.cpp
@@ -33,7 +33,7 @@ using namespace std;
 #include <srs_kernel_codec.hpp>
 #include <srs_kernel_rtp.hpp>
 #include <srs_app_hls.hpp>
-#include <srs_app_rtp.hpp>
+#include <srs_app_rtc.hpp>
 #include <srs_app_forward.hpp>
 #include <srs_app_config.hpp>
 #include <srs_app_encoder.hpp>
@@ -875,7 +875,7 @@ SrsOriginHub::SrsOriginHub()
     dash = new SrsDash();
     dvr = new SrsDvr();
     encoder = new SrsEncoder();
-    rtp = new SrsRtp();
+    rtc = new SrsRtc();
 #ifdef SRS_AUTO_HDS
     hds = new SrsHds();
 #endif
@@ -920,8 +920,8 @@ srs_error_t SrsOriginHub::initialize(SrsSource* s, SrsRequest* r)
         return srs_error_wrap(err, "format initialize");
     }
     
-    if ((err = rtp->initialize(this, req)) != srs_success) {
-        return srs_error_wrap(err, "rtp initialize");
+    if ((err = rtc->initialize(this, req)) != srs_success) {
+        return srs_error_wrap(err, "rtc initialize");
     }
     
     if ((err = hls->initialize(this, req)) != srs_success) {
@@ -1022,10 +1022,10 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
                   srs_flv_srates[c->sound_rate]);
     }
 
-    if ((err = rtp->on_audio(msg, format)) != srs_success) {
-        srs_warn("rtp: ignore audio error %s", srs_error_desc(err).c_str());
+    if ((err = rtc->on_audio(msg, format)) != srs_success) {
+        srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
         srs_error_reset(err);
-        rtp->on_unpublish();
+        rtc->on_unpublish();
     }
     
     if ((err = hls->on_audio(msg, format)) != srs_success) {
@@ -1122,11 +1122,11 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
     }
 
     // Parse RTMP message to RTP packets, in FU-A if too large.
-    if ((err = rtp->on_video(msg, format)) != srs_success) {
+    if ((err = rtc->on_video(msg, format)) != srs_success) {
         // TODO: We should support more strategies.
-        srs_warn("rtp: ignore video error %s", srs_error_desc(err).c_str());
+        srs_warn("rtc: ignore video error %s", srs_error_desc(err).c_str());
         srs_error_reset(err);
-        rtp->on_unpublish();
+        rtc->on_unpublish();
     }
 
     // TODO: FIXME: Refactor to move to rtp?
@@ -1201,8 +1201,8 @@ srs_error_t SrsOriginHub::on_publish()
         return srs_error_wrap(err, "encoder publish");
     }
     
-    if ((err = rtp->on_publish()) != srs_success) {
-        return srs_error_wrap(err, "rtp publish");
+    if ((err = rtc->on_publish()) != srs_success) {
+        return srs_error_wrap(err, "rtc publish");
     }
 
     if ((err = hls->on_publish()) != srs_success) {
@@ -1242,7 +1242,7 @@ void SrsOriginHub::on_unpublish()
     destroy_forwarders();
     
     encoder->on_unpublish();
-    rtp->on_unpublish();
+    rtc->on_unpublish();
     hls->on_unpublish();
     dash->on_unpublish();
     dvr->on_unpublish();
diff --git a/trunk/src/app/srs_app_source.hpp b/trunk/src/app/srs_app_source.hpp
index 083bf3e02..01005d96b 100644
--- a/trunk/src/app/srs_app_source.hpp
+++ b/trunk/src/app/srs_app_source.hpp
@@ -54,7 +54,7 @@ class SrsNgExec;
 class SrsConnection;
 class SrsMessageHeader;
 class SrsHls;
-class SrsRtp;
+class SrsRtc;
 class SrsDvr;
 class SrsDash;
 class SrsEncoder;
@@ -360,8 +360,8 @@ private:
 private:
     // The format, codec information.
     SrsRtmpFormat* format;
-    // rtp handler
-    SrsRtp* rtp;
+    // rtc handler
+    SrsRtc* rtc;
     // hls handler.
     SrsHls* hls;
     // The DASH encoder.